/*
 * Copyright (c) 2001-2003 The FFmpeg project
 *
 * first version by Francois Revol (revol@free.fr)
 * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
 *   by Mike Melanson (melanson@pcisys.net)
 * CD-ROM XA ADPCM codec by BERO
 * EA ADPCM decoder by Robin Kay (komadori@myrealbox.com)
 * EA ADPCM R1/R2/R3 decoder by Peter Ross (pross@xvid.org)
 * EA IMA EACS decoder by Peter Ross (pross@xvid.org)
 * EA IMA SEAD decoder by Peter Ross (pross@xvid.org)
 * EA ADPCM XAS decoder by Peter Ross (pross@xvid.org)
 * MAXIS EA ADPCM decoder by Robert Marston (rmarston@gmail.com)
 * THP ADPCM decoder by Marco Gerards (mgerards@xs4all.nl)
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include <inttypes.h>
#include <limits.h>
#include <stdlib.h>
#include <SDL2/SDL.h>

#include "../tiny_codec.h"
#include "../internal/avcodec.h"
#include "adpcm.h"
#include "adpcm_data.h"
#define BITSTREAM_READER_LE
#include "../internal/get_bits.h"
#include "../internal/bytestream.h"

/**
 * @file
 * ADPCM decoders
 * Features and limitations:
 *
 * Reference documents:
 * http://wiki.multimedia.cx/index.php?title=Category:ADPCM_Audio_Codecs
 * http://www.pcisys.net/~melanson/codecs/simpleaudio.html [dead]
 * http://www.geocities.com/SiliconValley/8682/aud3.txt [dead]
 * http://openquicktime.sourceforge.net/
 * XAnim sources (xa_codec.c) http://xanim.polter.net/
 * http://www.cs.ucla.edu/~leec/mediabench/applications.html [dead]
 * SoX source code http://sox.sourceforge.net/
 *
 * CD-ROM XA:
 * http://ku-www.ss.titech.ac.jp/~yatsushi/xaadpcm.html [dead]
 * vagpack & depack http://homepages.compuserve.de/bITmASTER32/psx-index.html [dead]
 * readstr http://www.geocities.co.jp/Playtown/2004/
 */

/* These are for CD-ROM XA ADPCM */
static const int xa_adpcm_table[5][2] = {
    {   0,   0 },
    {  60,   0 },
    { 115, -52 },
    {  98, -55 },
    { 122, -60 }
};

static const int ea_adpcm_table[] = {
    0,  240,  460,  392,
    0,    0, -208, -220,
    0,    1,    3,    4,
    7,    8,   10,   11,
    0,   -1,   -3,   -4
};

// padded to zero where table size is less then 16
static const int swf_index_tables[4][16] = {
    /*2*/ { -1, 2 },
    /*3*/ { -1, -1, 2, 4 },
    /*4*/ { -1, -1, -1, -1, 2, 4, 6, 8 },
    /*5*/ { -1, -1, -1, -1, -1, -1, -1, -1, 1, 2, 4, 6, 8, 10, 13, 16 }
};

/* end of tables */

typedef struct ADPCMDecodeContext
{
    ADPCMChannelStatus status[14];
    int vqa_version;                /**< VQA version. Used for ADPCM_IMA_WS */
    int has_status;
} ADPCMDecodeContext;

static inline int16_t adpcm_ima_expand_nibble(ADPCMChannelStatus *c, int8_t nibble, int shift)
{
    int step_index;
    int predictor;
    int sign, delta, diff, step;

    step = ff_adpcm_step_table[c->step_index];
    step_index = c->step_index + ff_adpcm_index_table[(unsigned)nibble];
    step_index = av_clip_c(step_index, 0, 88);

    sign = nibble & 8;
    delta = nibble & 7;
    /* perform direct multiplication instead of series of jumps proposed by
     * the reference ADPCM implementation since modern CPUs can do the mults
     * quickly enough */
    diff = ((2 * delta + 1) * step) >> shift;
    predictor = c->predictor;
    if (sign) predictor -= diff;
    else predictor += diff;

    c->predictor = av_clip_int16_c(predictor);
    c->step_index = step_index;

    return (int16_t)c->predictor;
}

static inline int16_t adpcm_ima_wav_expand_nibble(ADPCMChannelStatus *c, GetBitContext *gb, int bps)
{
    int nibble, step_index, predictor, sign, delta, diff, step, shift;

    shift = bps - 1;
    nibble = get_bits_le(gb, bps),
    step = ff_adpcm_step_table[c->step_index];
    step_index = c->step_index + ff_adpcm_index_tables[bps - 2][nibble];
    step_index = av_clip_c(step_index, 0, 88);

    sign = nibble & (1 << shift);
    delta = av_mod_uintp2_c(nibble, shift);
    diff = ((2 * delta + 1) * step) >> shift;
    predictor = c->predictor;
    if (sign) predictor -= diff;
    else predictor += diff;

    c->predictor = av_clip_int16_c(predictor);
    c->step_index = step_index;

    return (int16_t)c->predictor;
}

static inline int adpcm_ima_qt_expand_nibble(ADPCMChannelStatus *c, int nibble, int shift)
{
    int step_index;
    int predictor;
    int diff, step;

    step = ff_adpcm_step_table[c->step_index];
    step_index = c->step_index + ff_adpcm_index_table[nibble];
    step_index = av_clip_c(step_index, 0, 88);

    diff = step >> 3;
    if (nibble & 4) diff += step;
    if (nibble & 2) diff += step >> 1;
    if (nibble & 1) diff += step >> 2;

    if (nibble & 8)
        predictor = c->predictor - diff;
    else
        predictor = c->predictor + diff;

    c->predictor = av_clip_int16_c(predictor);
    c->step_index = step_index;

    return c->predictor;
}

static inline int16_t adpcm_ms_expand_nibble(ADPCMChannelStatus *c, int nibble)
{
    int predictor;

    predictor = (((c->sample1) * (c->coeff1)) + ((c->sample2) * (c->coeff2))) / 64;
    predictor += ((nibble & 0x08)?(nibble - 0x10):(nibble)) * c->idelta;

    c->sample2 = c->sample1;
    c->sample1 = av_clip_int16_c(predictor);
    c->idelta = (ff_adpcm_AdaptationTable[(int)nibble] * c->idelta) >> 8;
    if (c->idelta < 16) c->idelta = 16;
    if (c->idelta > INT_MAX/768)
    {
        //av_log(NULL, AV_LOG_WARNING, "idelta overflow\n");
        c->idelta = INT_MAX/768;
    }

    return c->sample1;
}

static inline int16_t adpcm_ima_oki_expand_nibble(ADPCMChannelStatus *c, int nibble)
{
    int step_index, predictor, sign, delta, diff, step;

    step = ff_adpcm_oki_step_table[c->step_index];
    step_index = c->step_index + ff_adpcm_index_table[(unsigned)nibble];
    step_index = av_clip_c(step_index, 0, 48);

    sign = nibble & 8;
    delta = nibble & 7;
    diff = ((2 * delta + 1) * step) >> 3;
    predictor = c->predictor;
    if (sign) predictor -= diff;
    else predictor += diff;

    c->predictor = av_clip_intp2_c(predictor, 11);
    c->step_index = step_index;

    return c->predictor << 4;
}

static inline int16_t adpcm_ct_expand_nibble(ADPCMChannelStatus *c, int8_t nibble)
{
    int sign, delta, diff;
    int new_step;

    sign = nibble & 8;
    delta = nibble & 7;
    /* perform direct multiplication instead of series of jumps proposed by
     * the reference ADPCM implementation since modern CPUs can do the mults
     * quickly enough */
    diff = ((2 * delta + 1) * c->step) >> 3;
    /* predictor update is not so trivial: predictor is multiplied on 254/256 before updating */
    c->predictor = ((c->predictor * 254) >> 8) + (sign ? -diff : diff);
    c->predictor = av_clip_int16_c(c->predictor);
    /* calculate new step and clamp it to range 511..32767 */
    new_step = (ff_adpcm_AdaptationTable[nibble & 7] * c->step) >> 8;
    c->step = av_clip_c(new_step, 511, 32767);

    return (int16_t)c->predictor;
}

static inline int16_t adpcm_sbpro_expand_nibble(ADPCMChannelStatus *c, int8_t nibble, int size, int shift)
{
    int sign, delta, diff;

    sign = nibble & (1<<(size-1));
    delta = nibble & ((1<<(size-1))-1);
    diff = delta << (7 + c->step + shift);

    /* clamp result */
    c->predictor = av_clip_c(c->predictor + (sign ? -diff : diff), -16384, 16256);

    /* calculate new step */
    if (delta >= (2*size - 3) && c->step < 3)
        c->step++;
    else if (delta == 0 && c->step > 0)
        c->step--;

    return (int16_t) c->predictor;
}

static inline int16_t adpcm_yamaha_expand_nibble(ADPCMChannelStatus *c, uint8_t nibble)
{
    if(!c->step)
    {
        c->predictor = 0;
        c->step = 127;
    }

    c->predictor += (c->step * ff_adpcm_yamaha_difflookup[nibble]) / 8;
    c->predictor = av_clip_int16_c(c->predictor);
    c->step = (c->step * ff_adpcm_yamaha_indexscale[nibble]) >> 8;
    c->step = av_clip_c(c->step, 127, 24576);
    return c->predictor;
}

static inline int16_t adpcm_mtaf_expand_nibble(ADPCMChannelStatus *c, uint8_t nibble)
{
    c->predictor += ff_adpcm_mtaf_stepsize[c->step][nibble];
    c->predictor = av_clip_int16_c(c->predictor);
    c->step += ff_adpcm_index_table[nibble];
    c->step = av_clip_uintp2_c(c->step, 5);
    return c->predictor;
}

static int xa_decode(struct tiny_codec_s *avctx, int16_t *out0, int16_t *out1,
                     const uint8_t *in, ADPCMChannelStatus *left,
                     ADPCMChannelStatus *right, int channels, int sample_offset)
{
    int i, j;
    int shift,filter,f0,f1;
    int s_1,s_2;
    int d,s,t;

    out0 += sample_offset;
    if (channels == 1)
        out1 = out0 + 28;
    else
        out1 += sample_offset;

    for(i=0;i<4;i++)
    {
        shift  = 12 - (in[4+i*2] & 15);
        filter = in[4+i*2] >> 4;
        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table))
        {
            //avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
            filter=0;
        }
        f0 = xa_adpcm_table[filter][0];
        f1 = xa_adpcm_table[filter][1];

        s_1 = left->sample1;
        s_2 = left->sample2;

        for(j=0;j<28;j++)
        {
            d = in[16+i+j*4];

            t = sign_extend(d, 4);
            s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
            s_2 = s_1;
            s_1 = av_clip_int16_c(s);
            out0[j] = s_1;
        }

        if (channels == 2)
        {
            left->sample1 = s_1;
            left->sample2 = s_2;
            s_1 = right->sample1;
            s_2 = right->sample2;
        }

        shift  = 12 - (in[5+i*2] & 15);
        filter = in[5+i*2] >> 4;
        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table))
        {
            //avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
            filter=0;
        }

        f0 = xa_adpcm_table[filter][0];
        f1 = xa_adpcm_table[filter][1];

        for(j=0;j<28;j++)
        {
            d = in[16+i+j*4];

            t = sign_extend(d >> 4, 4);
            s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
            s_2 = s_1;
            s_1 = av_clip_int16_c(s);
            out1[j] = s_1;
        }

        if (channels == 2)
        {
            right->sample1 = s_1;
            right->sample2 = s_2;
        }
        else
        {
            left->sample1 = s_1;
            left->sample2 = s_2;
        }

        out0 += 28 * (3 - channels);
        out1 += 28 * (3 - channels);
    }

    return 0;
}

static void adpcm_swf_decode(struct tiny_codec_s *avctx, const uint8_t *buf, int buf_size, int16_t *samples)
{
    ADPCMDecodeContext *c = (ADPCMDecodeContext*)avctx->audio.priv_data;
    GetBitContext gb;
    const int *table;
    int k0, signmask, nb_bits, count;
    int size = buf_size*8;
    int i;

    init_get_bits(&gb, buf, size);

    //read bits & initial values
    nb_bits = get_bits(&gb, 2)+2;
    table = swf_index_tables[nb_bits-2];
    k0 = 1 << (nb_bits-2);
    signmask = 1 << (nb_bits-1);

    while (get_bits_count(&gb) <= size - 22 * avctx->audio.channels)
    {
        for (i = 0; i < avctx->audio.channels; i++)
        {
            *samples++ = c->status[i].predictor = get_sbits(&gb, 16);
            c->status[i].step_index = get_bits(&gb, 6);
        }

        for (count = 0; get_bits_count(&gb) <= size - nb_bits*avctx->audio.channels && count < 4095; count++)
        {
            for (int i = 0; i < avctx->audio.channels; i++)
            {
                // similar to IMA adpcm
                int delta = get_bits(&gb, nb_bits);
                int step = ff_adpcm_step_table[c->status[i].step_index];
                int vpdiff = 0; // vpdiff = (delta+0.5)*step/4
                int k = k0;

                do
                {
                    if (delta & k)
                        vpdiff += step;
                    step >>= 1;
                    k >>= 1;
                } while(k);
                vpdiff += step;

                if (delta & signmask)
                    c->status[i].predictor -= vpdiff;
                else
                    c->status[i].predictor += vpdiff;

                c->status[i].step_index += table[delta & (~signmask)];

                c->status[i].step_index = av_clip_c(c->status[i].step_index, 0, 88);
                c->status[i].predictor = av_clip_int16_c(c->status[i].predictor);

                *samples++ = c->status[i].predictor;
            }
        }
    }
}

/**
 * Get the number of samples that will be decoded from the packet.
 * In one case, this is actually the maximum number of samples possible to
 * decode with the given buf_size.
 *
 * @param[out] coded_samples set to the number of samples as coded in the
 *                           packet, or 0 if the codec does not encode the
 *                           number of samples in each frame.
 * @param[out] approx_nb_samples set to non-zero if the number of samples
 *                               returned is an approximation.
 */
static int get_nb_samples(struct tiny_codec_s *avctx, GetByteContext *gb,
                          int buf_size, int *coded_samples, int *approx_nb_samples)
{
    ADPCMDecodeContext *s = (ADPCMDecodeContext*)avctx->audio.priv_data;
    int nb_samples        = 0;
    int ch                = avctx->audio.channels;
    int has_coded_samples = 0;
    int header_size;

    *coded_samples = 0;
    *approx_nb_samples = 0;

    if(ch <= 0)
        return 0;

    switch (avctx->audio.codec_tag)
    {
        /* constant, only check buf_size */
        case AV_CODEC_ID_ADPCM_EA_XAS:
            if (buf_size < 76 * ch)
                return 0;
            nb_samples = 128;
            break;
        case AV_CODEC_ID_ADPCM_IMA_QT:
            if (buf_size < 34 * ch)
                return 0;
            nb_samples = 64;
            break;
        /* simple 4-bit adpcm */
        case AV_CODEC_ID_ADPCM_CT:
        case AV_CODEC_ID_ADPCM_IMA_APC:
        case AV_CODEC_ID_ADPCM_IMA_EA_SEAD:
        case AV_CODEC_ID_ADPCM_IMA_OKI:
        case AV_CODEC_ID_ADPCM_IMA_WS:
        case AV_CODEC_ID_ADPCM_YAMAHA:
        case AV_CODEC_ID_ADPCM_AICA:
            nb_samples = buf_size * 2 / ch;
            break;
    }
    if (nb_samples)
        return nb_samples;

    /* simple 4-bit adpcm, with header */
    header_size = 0;
    switch (avctx->audio.codec_tag)
    {
        case AV_CODEC_ID_ADPCM_4XM:
        case AV_CODEC_ID_ADPCM_IMA_DAT4:
        case AV_CODEC_ID_ADPCM_IMA_ISS:     header_size = 4 * ch;      break;
        case AV_CODEC_ID_ADPCM_IMA_AMV:     header_size = 8;           break;
        case AV_CODEC_ID_ADPCM_IMA_SMJPEG:  header_size = 4 * ch;      break;
    }
    if (header_size > 0)
        return (buf_size - header_size) * 2 / ch;

    /* more complex formats */
    switch (avctx->audio.codec_tag)
    {
        case AV_CODEC_ID_ADPCM_EA:
            has_coded_samples = 1;
            *coded_samples  = bytestream2_get_le32(gb);
            *coded_samples -= *coded_samples % 28;
            nb_samples      = (buf_size - 12) / 30 * 28;
            break;
        case AV_CODEC_ID_ADPCM_IMA_EA_EACS:
            has_coded_samples = 1;
            *coded_samples = bytestream2_get_le32(gb);
            nb_samples     = (buf_size - (4 + 8 * ch)) * 2 / ch;
            break;
        case AV_CODEC_ID_ADPCM_EA_MAXIS_XA:
            nb_samples = (buf_size - ch) / ch * 2;
            break;
        case AV_CODEC_ID_ADPCM_EA_R1:
        case AV_CODEC_ID_ADPCM_EA_R2:
        case AV_CODEC_ID_ADPCM_EA_R3:
            /* maximum number of samples */
            /* has internal offsets and a per-frame switch to signal raw 16-bit */
            has_coded_samples = 1;
            switch (avctx->audio.codec_tag)
            {
                case AV_CODEC_ID_ADPCM_EA_R1:
                    header_size    = 4 + 9 * ch;
                    *coded_samples = bytestream2_get_le32(gb);
                    break;
                case AV_CODEC_ID_ADPCM_EA_R2:
                    header_size    = 4 + 5 * ch;
                    *coded_samples = bytestream2_get_le32(gb);
                    break;
                case AV_CODEC_ID_ADPCM_EA_R3:
                    header_size    = 4 + 5 * ch;
                    *coded_samples = bytestream2_get_be32(gb);
                    break;
            }
            *coded_samples -= *coded_samples % 28;
            nb_samples      = (buf_size - header_size) * 2 / ch;
            nb_samples     -= nb_samples % 28;
            *approx_nb_samples = 1;
            break;
        case AV_CODEC_ID_ADPCM_IMA_DK3:
            if (avctx->audio.block_align > 0)
                buf_size = FFMIN(buf_size, avctx->audio.block_align);
            nb_samples = ((buf_size - 16) * 2 / 3 * 4) / ch;
            break;
        case AV_CODEC_ID_ADPCM_IMA_DK4:
            if (avctx->audio.block_align > 0)
                buf_size = FFMIN(buf_size, avctx->audio.block_align);
            if (buf_size < 4 * ch)
                return -1;
            nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch;
            break;
        case AV_CODEC_ID_ADPCM_IMA_RAD:
            if (avctx->audio.block_align > 0)
                buf_size = FFMIN(buf_size, avctx->audio.block_align);
            nb_samples = (buf_size - 4 * ch) * 2 / ch;
            break;
        case AV_CODEC_ID_ADPCM_IMA_WAV:
        {
            int bsize = ff_adpcm_ima_block_sizes[avctx->audio.bits_per_coded_sample - 2];
            int bsamples = ff_adpcm_ima_block_samples[avctx->audio.bits_per_coded_sample - 2];
            if (avctx->audio.block_align > 0)
                buf_size = FFMIN(buf_size, avctx->audio.block_align);
            if (buf_size < 4 * ch)
                return -1;
            nb_samples = 1 + (buf_size - 4 * ch) / (bsize * ch) * bsamples;
            break;
        }
        case AV_CODEC_ID_ADPCM_MS:
            if (avctx->audio.block_align > 0)
                buf_size = FFMIN(buf_size, avctx->audio.block_align);
            nb_samples = (buf_size - 6 * ch) * 2 / ch;
            break;
        case AV_CODEC_ID_ADPCM_MTAF:
            if (avctx->audio.block_align > 0)
                buf_size = FFMIN(buf_size, avctx->audio.block_align);
            nb_samples = (buf_size - 16 * (ch / 2)) * 2 / ch;
            break;
        case AV_CODEC_ID_ADPCM_SBPRO_2:
        case AV_CODEC_ID_ADPCM_SBPRO_3:
        case AV_CODEC_ID_ADPCM_SBPRO_4:
        {
            int samples_per_byte;
            switch (avctx->audio.codec_tag)
            {
                case AV_CODEC_ID_ADPCM_SBPRO_2: samples_per_byte = 4; break;
                case AV_CODEC_ID_ADPCM_SBPRO_3: samples_per_byte = 3; break;
                case AV_CODEC_ID_ADPCM_SBPRO_4: samples_per_byte = 2; break;
            }
            if (!s->status[0].step_index)
            {
                if (buf_size < ch)
                    return -1;
                nb_samples++;
                buf_size -= ch;
            }
            nb_samples += buf_size * samples_per_byte / ch;
            break;
        }
        case AV_CODEC_ID_ADPCM_SWF:
        {
            int buf_bits       = buf_size * 8 - 2;
            int nbits          = (bytestream2_get_byte(gb) >> 6) + 2;
            int block_hdr_size = 22 * ch;
            int block_size     = block_hdr_size + nbits * ch * 4095;
            int nblocks        = buf_bits / block_size;
            int bits_left      = buf_bits - nblocks * block_size;
            nb_samples         = nblocks * 4096;
            if (bits_left >= block_hdr_size)
                nb_samples += 1 + (bits_left - block_hdr_size) / (nbits * ch);
            break;
        }
        case AV_CODEC_ID_ADPCM_THP:
        case AV_CODEC_ID_ADPCM_THP_LE:
            if (avctx->audio.extradata)
            {
                nb_samples = buf_size * 14 / (8 * ch);
                break;
            }
            has_coded_samples = 1;
            bytestream2_skip(gb, 4); // channel size
            *coded_samples  = (avctx->audio.codec_tag == AV_CODEC_ID_ADPCM_THP_LE) ?
                              bytestream2_get_le32(gb) :
                              bytestream2_get_be32(gb);
            buf_size       -= 8 + 36 * ch;
            buf_size       /= ch;
            nb_samples      = buf_size / 8 * 14;
            if (buf_size % 8 > 1)
                nb_samples     += (buf_size % 8 - 1) * 2;
            *approx_nb_samples = 1;
            break;
        case AV_CODEC_ID_ADPCM_AFC:
            nb_samples = buf_size / (9 * ch) * 16;
            break;
        case AV_CODEC_ID_ADPCM_XA:
            nb_samples = (buf_size / 128) * 224 / ch;
            break;
        case AV_CODEC_ID_ADPCM_DTK:
        case AV_CODEC_ID_ADPCM_PSX:
            nb_samples = buf_size / (16 * ch) * 28;
            break;
    }

    /* validate coded sample count */
    if (has_coded_samples && (*coded_samples <= 0 || *coded_samples > nb_samples))
        return -1;

    return nb_samples;
}


static int32_t adpcm_decode_frame(struct tiny_codec_s *avctx, struct AVPacket *avpkt)
{
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    ADPCMDecodeContext *c = (ADPCMDecodeContext*)avctx->audio.priv_data;
    ADPCMChannelStatus *cs;
    int n, m, channel, i;
    int16_t *samples;
    int16_t **samples_p;
    uint32_t output_buff_size = 0;
    int st; /* stereo */
    int count1, count2;
    int nb_samples, coded_samples, approx_nb_samples, ret;
    GetByteContext gb;

    bytestream2_init(&gb, buf, buf_size);
    nb_samples = get_nb_samples(avctx, &gb, buf_size, &coded_samples, &approx_nb_samples);
    if (nb_samples <= 0)
    {
        //av_log(avctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
        return -1;
    }

    /* get output buffer */
    if(coded_samples)
    {
        nb_samples = coded_samples;
    }
    output_buff_size = codec_resize_audio_buffer(avctx, avctx->audio.bits_per_sample >> 3, nb_samples);
    samples = (int16_t*)avctx->audio.buff;
    samples_p = (int16_t**)avctx->audio.buff_p;

    st = avctx->audio.channels == 2 ? 1 : 0;

    switch(avctx->audio.codec_tag)
    {
        case AV_CODEC_ID_ADPCM_IMA_QT:
            /* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples).
               Channel data is interleaved per-chunk. */
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                int predictor;
                int step_index;
                cs = &(c->status[channel]);
                /* (pppppp) (piiiiiii) */

                /* Bits 15-7 are the _top_ 9 bits of the 16-bit initial predictor value */
                predictor = sign_extend(bytestream2_get_be16u(&gb), 16);
                step_index = predictor & 0x7F;
                predictor &= ~0x7F;

                if (cs->step_index == step_index)
                {
                    int diff = predictor - cs->predictor;
                    if (diff < 0)
                        diff = - diff;
                    if (diff > 0x7f)
                        goto update;
                }
                else
                {
                update:
                    cs->step_index = step_index;
                    cs->predictor = predictor;
                }

                if (cs->step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //       channel, cs->step_index);
                    return -1;
                }

                samples = samples_p[channel];

                for (m = 0; m < 64; m += 2)
                {
                    int byte = bytestream2_get_byteu(&gb);
                    samples[m    ] = adpcm_ima_qt_expand_nibble(cs, byte & 0x0F, 3);
                    samples[m + 1] = adpcm_ima_qt_expand_nibble(cs, byte >> 4  , 3);
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_WAV:
            for(i=0; i<avctx->audio.channels; i++)
            {
                cs = &(c->status[i]);
                cs->predictor = samples_p[i][0] = sign_extend(bytestream2_get_le16u(&gb), 16);

                cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
                if (cs->step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //       i, cs->step_index);
                    return -1;
                }
            }

            if (avctx->audio.bits_per_coded_sample != 4)
            {
                int samples_per_block = ff_adpcm_ima_block_samples[avctx->audio.bits_per_coded_sample - 2];
                int block_size = ff_adpcm_ima_block_sizes[avctx->audio.bits_per_coded_sample - 2];
                uint8_t temp[20 + AV_INPUT_BUFFER_PADDING_SIZE] = { 0 };
                GetBitContext g;

                for (n = 0; n < (nb_samples - 1) / samples_per_block; n++)
                {
                    for (i = 0; i < avctx->audio.channels; i++)
                    {
                        int j;

                        cs = &c->status[i];
                        samples = &samples_p[i][1 + n * samples_per_block];
                        for (j = 0; j < block_size; j++)
                        {
                            temp[j] = buf[4 * avctx->audio.channels + block_size * n * avctx->audio.channels +
                                            (j % 4) + (j / 4) * (avctx->audio.channels * 4) + i * 4];
                        }
                        ret = init_get_bits8(&g, (const uint8_t *)&temp, block_size);
                        if (ret < 0)
                            return ret;
                        for (m = 0; m < samples_per_block; m++)
                        {
                            samples[m] = adpcm_ima_wav_expand_nibble(cs, &g,
                                              avctx->audio.bits_per_coded_sample);
                        }
                    }
                }
                bytestream2_skip(&gb, avctx->audio.block_align - avctx->audio.channels * 4);
            }
            else
            {
                for (n = 0; n < (nb_samples - 1) / 8; n++)
                {
                    for (i = 0; i < avctx->audio.channels; i++)
                    {
                        cs = &c->status[i];
                        samples = &samples_p[i][1 + n * 8];
                        for (m = 0; m < 8; m += 2)
                        {
                            int v = bytestream2_get_byteu(&gb);
                            samples[m    ] = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
                            samples[m + 1] = adpcm_ima_expand_nibble(cs, v >> 4  , 3);
                        }
                    }
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_4XM:
            for (i = 0; i < avctx->audio.channels; i++)
                c->status[i].predictor = sign_extend(bytestream2_get_le16u(&gb), 16);

            for (i = 0; i < avctx->audio.channels; i++)
            {
                c->status[i].step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
                if (c->status[i].step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //       i, c->status[i].step_index);
                    return -1;
                }
            }

            for (i = 0; i < avctx->audio.channels; i++)
            {
                samples = (int16_t*)avctx->audio.buff_p[i];
                cs = &c->status[i];
                for (n = nb_samples >> 1; n > 0; n--)
                {
                    int v = bytestream2_get_byteu(&gb);
                    *samples++ = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
                    *samples++ = adpcm_ima_expand_nibble(cs, v >> 4  , 4);
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_MS:
        {
            int block_predictor;

            block_predictor = bytestream2_get_byteu(&gb);
            if (block_predictor > 6)
            {
                //av_log(avctx, AV_LOG_ERROR, "ERROR: block_predictor[0] = %d\n",
                //       block_predictor);
                return -1;
            }
            c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor];
            c->status[0].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor];
            if (st)
            {
                block_predictor = bytestream2_get_byteu(&gb);
                if (block_predictor > 6)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: block_predictor[1] = %d\n",
                    //       block_predictor);
                    return -1;
                }
                c->status[1].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor];
                c->status[1].coeff2 = ff_adpcm_AdaptCoeff2[block_predictor];
            }
            c->status[0].idelta = sign_extend(bytestream2_get_le16u(&gb), 16);
            if (st)
            {
                c->status[1].idelta = sign_extend(bytestream2_get_le16u(&gb), 16);
            }

            c->status[0].sample1 = sign_extend(bytestream2_get_le16u(&gb), 16);
            if (st) c->status[1].sample1 = sign_extend(bytestream2_get_le16u(&gb), 16);
            c->status[0].sample2 = sign_extend(bytestream2_get_le16u(&gb), 16);
            if (st) c->status[1].sample2 = sign_extend(bytestream2_get_le16u(&gb), 16);

            *samples++ = c->status[0].sample2;
            if (st) *samples++ = c->status[1].sample2;
            *samples++ = c->status[0].sample1;
            if (st) *samples++ = c->status[1].sample1;
            for(n = (nb_samples - 2) >> (1 - st); n > 0; n--)
            {
                int byte = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ms_expand_nibble(&c->status[0 ], byte >> 4  );
                *samples++ = adpcm_ms_expand_nibble(&c->status[st], byte & 0x0F);
            }
            break;
        }
        case AV_CODEC_ID_ADPCM_MTAF:
            for (channel = 0; channel < avctx->audio.channels; channel+=2)
            {
                bytestream2_skipu(&gb, 4);
                c->status[channel    ].step      = bytestream2_get_le16u(&gb) & 0x1f;
                c->status[channel + 1].step      = bytestream2_get_le16u(&gb) & 0x1f;
                c->status[channel    ].predictor = sign_extend(bytestream2_get_le16u(&gb), 16);
                bytestream2_skipu(&gb, 2);
                c->status[channel + 1].predictor = sign_extend(bytestream2_get_le16u(&gb), 16);
                bytestream2_skipu(&gb, 2);
                for (n = 0; n < nb_samples; n+=2)
                {
                    int v = bytestream2_get_byteu(&gb);
                    samples_p[channel][n    ] = adpcm_mtaf_expand_nibble(&c->status[channel], v & 0x0F);
                    samples_p[channel][n + 1] = adpcm_mtaf_expand_nibble(&c->status[channel], v >> 4  );
                }
                for (n = 0; n < nb_samples; n+=2)
                {
                    int v = bytestream2_get_byteu(&gb);
                    samples_p[channel + 1][n    ] = adpcm_mtaf_expand_nibble(&c->status[channel + 1], v & 0x0F);
                    samples_p[channel + 1][n + 1] = adpcm_mtaf_expand_nibble(&c->status[channel + 1], v >> 4  );
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_DK4:
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                cs = &c->status[channel];
                cs->predictor  = *samples++ = sign_extend(bytestream2_get_le16u(&gb), 16);
                cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
                if (cs->step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //       channel, cs->step_index);
                    return -1;
                }
            }
            for (n = (nb_samples - 1) >> (1 - st); n > 0; n--)
            {
                int v = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4  , 3);
                *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_DK3:
        {
            int last_byte = 0;
            int nibble;
            int decode_top_nibble_next = 0;
            int diff_channel;
            const int16_t *samples_end = samples + avctx->audio.channels * nb_samples;

            bytestream2_skipu(&gb, 10);
            c->status[0].predictor  = sign_extend(bytestream2_get_le16u(&gb), 16);
            c->status[1].predictor  = sign_extend(bytestream2_get_le16u(&gb), 16);
            c->status[0].step_index = bytestream2_get_byteu(&gb);
            c->status[1].step_index = bytestream2_get_byteu(&gb);
            if (c->status[0].step_index > 88u || c->status[1].step_index > 88u)
            {
                //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i/%i\n",
                //       c->status[0].step_index, c->status[1].step_index);
                return -1;
            }
            /* sign extend the predictors */
            diff_channel = c->status[1].predictor;

            /* DK3 ADPCM support macro */
    #define DK3_GET_NEXT_NIBBLE() \
        if (decode_top_nibble_next) { \
            nibble = last_byte >> 4; \
            decode_top_nibble_next = 0; \
        } else { \
            last_byte = bytestream2_get_byteu(&gb); \
            nibble = last_byte & 0x0F; \
            decode_top_nibble_next = 1; \
        }

            while (samples < samples_end)
            {

                /* for this algorithm, c->status[0] is the sum channel and
                 * c->status[1] is the diff channel */

                /* process the first predictor of the sum channel */
                DK3_GET_NEXT_NIBBLE();
                adpcm_ima_expand_nibble(&c->status[0], nibble, 3);

                /* process the diff channel predictor */
                DK3_GET_NEXT_NIBBLE();
                adpcm_ima_expand_nibble(&c->status[1], nibble, 3);

                /* process the first pair of stereo PCM samples */
                diff_channel = (diff_channel + c->status[1].predictor) / 2;
                *samples++ = c->status[0].predictor + c->status[1].predictor;
                *samples++ = c->status[0].predictor - c->status[1].predictor;

                /* process the second predictor of the sum channel */
                DK3_GET_NEXT_NIBBLE();
                adpcm_ima_expand_nibble(&c->status[0], nibble, 3);

                /* process the second pair of stereo PCM samples */
                diff_channel = (diff_channel + c->status[1].predictor) / 2;
                *samples++ = c->status[0].predictor + c->status[1].predictor;
                *samples++ = c->status[0].predictor - c->status[1].predictor;
            }

            if ((bytestream2_tell(&gb) & 1))
                bytestream2_skip(&gb, 1);
            break;
        }
        case AV_CODEC_ID_ADPCM_IMA_ISS:
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                cs = &c->status[channel];
                cs->predictor  = sign_extend(bytestream2_get_le16u(&gb), 16);
                cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
                if (cs->step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //       channel, cs->step_index);
                    return -1;
                }
            }

            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int v1, v2;
                int v = bytestream2_get_byteu(&gb);
                /* nibbles are swapped for mono */
                if (st)
                {
                    v1 = v >> 4;
                    v2 = v & 0x0F;
                }
                else
                {
                    v2 = v >> 4;
                    v1 = v & 0x0F;
                }
                *samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v1, 3);
                *samples++ = adpcm_ima_expand_nibble(&c->status[st], v2, 3);
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_DAT4:
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                cs = &c->status[channel];
                samples = samples_p[channel];
                bytestream2_skip(&gb, 4);
                for (n = 0; n < nb_samples; n += 2)
                {
                    int v = bytestream2_get_byteu(&gb);
                    *samples++ = adpcm_ima_expand_nibble(cs, v >> 4  , 3);
                    *samples++ = adpcm_ima_expand_nibble(cs, v & 0x0F, 3);
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_APC:
            while (bytestream2_get_bytes_left(&gb) > 0)
            {
                int v = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ima_expand_nibble(&c->status[0],  v >> 4  , 3);
                *samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_OKI:
            while (bytestream2_get_bytes_left(&gb) > 0)
            {
                int v = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ima_oki_expand_nibble(&c->status[0],  v >> 4  );
                *samples++ = adpcm_ima_oki_expand_nibble(&c->status[st], v & 0x0F);
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_RAD:
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                cs = &c->status[channel];
                cs->step_index = sign_extend(bytestream2_get_le16u(&gb), 16);
                cs->predictor  = sign_extend(bytestream2_get_le16u(&gb), 16);
                if (cs->step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //       channel, cs->step_index);
                    return -1;
                }
            }
            for (n = 0; n < nb_samples / 2; n++)
            {
                int byte[2];

                byte[0] = bytestream2_get_byteu(&gb);
                if (st)
                    byte[1] = bytestream2_get_byteu(&gb);
                for(channel = 0; channel < avctx->audio.channels; channel++)
                {
                    *samples++ = adpcm_ima_expand_nibble(&c->status[channel], byte[channel] & 0x0F, 3);
                }
                for(channel = 0; channel < avctx->audio.channels; channel++)
                {
                    *samples++ = adpcm_ima_expand_nibble(&c->status[channel], byte[channel] >> 4  , 3);
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_WS:
            if (c->vqa_version == 3)
            {
                for (channel = 0; channel < avctx->audio.channels; channel++)
                {
                    int16_t *smp = samples_p[channel];

                    for (n = nb_samples / 2; n > 0; n--)
                    {
                        int v = bytestream2_get_byteu(&gb);
                        *smp++ = adpcm_ima_expand_nibble(&c->status[channel], v >> 4  , 3);
                        *smp++ = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
                    }
                }
            }
            else
            {
                for (n = nb_samples / 2; n > 0; n--)
                {
                    for (channel = 0; channel < avctx->audio.channels; channel++)
                    {
                        int v = bytestream2_get_byteu(&gb);
                        *samples++  = adpcm_ima_expand_nibble(&c->status[channel], v >> 4  , 3);
                        samples[st] = adpcm_ima_expand_nibble(&c->status[channel], v & 0x0F, 3);
                    }
                    samples += avctx->audio.channels;
                }
            }
            bytestream2_seek(&gb, 0, SEEK_END);
            break;
        case AV_CODEC_ID_ADPCM_XA:
        {
            int16_t *out0 = samples_p[0];
            int16_t *out1 = samples_p[1];
            int samples_per_block = 28 * (3 - avctx->audio.channels) * 4;
            int sample_offset = 0;
            while (bytestream2_get_bytes_left(&gb) >= 128)
            {
                if ((ret = xa_decode(avctx, out0, out1, buf + bytestream2_tell(&gb),
                                     &c->status[0], &c->status[1],
                                     avctx->audio.channels, sample_offset)) < 0)
                    return ret;
                bytestream2_skipu(&gb, 128);
                sample_offset += samples_per_block;
            }
            break;
        }
        case AV_CODEC_ID_ADPCM_IMA_EA_EACS:
            for (i=0; i<=st; i++)
            {
                c->status[i].step_index = bytestream2_get_le32u(&gb);
                if (c->status[i].step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index[%d] = %i\n",
                    //      i, c->status[i].step_index);
                    return -1;
                }
            }
            for (i=0; i<=st; i++)
                c->status[i].predictor  = bytestream2_get_le32u(&gb);

            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int byte   = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ima_expand_nibble(&c->status[0],  byte >> 4,   3);
                *samples++ = adpcm_ima_expand_nibble(&c->status[st], byte & 0x0F, 3);
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_EA_SEAD:
            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int byte = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ima_expand_nibble(&c->status[0],  byte >> 4,   6);
                *samples++ = adpcm_ima_expand_nibble(&c->status[st], byte & 0x0F, 6);
            }
            break;
        case AV_CODEC_ID_ADPCM_EA:
        {
            int previous_left_sample, previous_right_sample;
            int current_left_sample, current_right_sample;
            int next_left_sample, next_right_sample;
            int coeff1l, coeff2l, coeff1r, coeff2r;
            int shift_left, shift_right;

            /* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces,
               each coding 28 stereo samples. */

            if(avctx->audio.channels != 2)
                return -1;

            current_left_sample   = sign_extend(bytestream2_get_le16u(&gb), 16);
            previous_left_sample  = sign_extend(bytestream2_get_le16u(&gb), 16);
            current_right_sample  = sign_extend(bytestream2_get_le16u(&gb), 16);
            previous_right_sample = sign_extend(bytestream2_get_le16u(&gb), 16);

            for (count1 = 0; count1 < nb_samples / 28; count1++)
            {
                int byte = bytestream2_get_byteu(&gb);
                coeff1l = ea_adpcm_table[ byte >> 4       ];
                coeff2l = ea_adpcm_table[(byte >> 4  ) + 4];
                coeff1r = ea_adpcm_table[ byte & 0x0F];
                coeff2r = ea_adpcm_table[(byte & 0x0F) + 4];

                byte = bytestream2_get_byteu(&gb);
                shift_left  = 20 - (byte >> 4);
                shift_right = 20 - (byte & 0x0F);

                for (count2 = 0; count2 < 28; count2++)
                {
                    byte = bytestream2_get_byteu(&gb);
                    next_left_sample  = sign_extend(byte >> 4, 4) << shift_left;
                    next_right_sample = sign_extend(byte,      4) << shift_right;

                    next_left_sample = (next_left_sample +
                        (current_left_sample * coeff1l) +
                        (previous_left_sample * coeff2l) + 0x80) >> 8;
                    next_right_sample = (next_right_sample +
                        (current_right_sample * coeff1r) +
                        (previous_right_sample * coeff2r) + 0x80) >> 8;

                    previous_left_sample = current_left_sample;
                    current_left_sample = av_clip_int16_c(next_left_sample);
                    previous_right_sample = current_right_sample;
                    current_right_sample = av_clip_int16_c(next_right_sample);
                    *samples++ = current_left_sample;
                    *samples++ = current_right_sample;
                }
            }

            bytestream2_skip(&gb, 2); // Skip terminating 0x0000

            break;
        }
        case AV_CODEC_ID_ADPCM_EA_MAXIS_XA:
        {
            int coeff[2][2], shift[2];

            for(channel = 0; channel < avctx->audio.channels; channel++)
            {
                int byte = bytestream2_get_byteu(&gb);
                for (i=0; i<2; i++)
                    coeff[channel][i] = ea_adpcm_table[(byte >> 4) + 4*i];
                shift[channel] = 20 - (byte & 0x0F);
            }
            for (count1 = 0; count1 < nb_samples / 2; count1++)
            {
                int byte[2];

                byte[0] = bytestream2_get_byteu(&gb);
                if (st) byte[1] = bytestream2_get_byteu(&gb);
                for(i = 4; i >= 0; i-=4) /* Pairwise samples LL RR (st) or LL LL (mono) */
                {
                    for(channel = 0; channel < avctx->audio.channels; channel++)
                    {
                        int sample = sign_extend(byte[channel] >> i, 4) << shift[channel];
                        sample = (sample +
                                 c->status[channel].sample1 * coeff[channel][0] +
                                 c->status[channel].sample2 * coeff[channel][1] + 0x80) >> 8;
                        c->status[channel].sample2 = c->status[channel].sample1;
                        c->status[channel].sample1 = av_clip_int16_c(sample);
                        *samples++ = c->status[channel].sample1;
                    }
                }
            }
            bytestream2_seek(&gb, 0, SEEK_END);
            break;
        }
        case AV_CODEC_ID_ADPCM_EA_R1:
        case AV_CODEC_ID_ADPCM_EA_R2:
        case AV_CODEC_ID_ADPCM_EA_R3:
        {
            /* channel numbering
               2chan: 0=fl, 1=fr
               4chan: 0=fl, 1=rl, 2=fr, 3=rr
               6chan: 0=fl, 1=c,  2=fr, 3=rl,  4=rr, 5=sub */
            const int big_endian = avctx->audio.codec_tag == AV_CODEC_ID_ADPCM_EA_R3;
            int previous_sample, current_sample, next_sample;
            int coeff1, coeff2;
            int shift;
            unsigned int channel;
            int16_t *samplesC;
            int count = 0;
            int offsets[6];

            for (channel=0; channel<avctx->audio.channels; channel++)
                offsets[channel] = (big_endian ? bytestream2_get_be32(&gb) :
                                                 bytestream2_get_le32(&gb)) +
                                   (avctx->audio.channels + 1) * 4;

            for (channel=0; channel<avctx->audio.channels; channel++)
            {
                bytestream2_seek(&gb, offsets[channel], SEEK_SET);
                samplesC = samples_p[channel];

                if (avctx->audio.codec_tag == AV_CODEC_ID_ADPCM_EA_R1)
                {
                    current_sample  = sign_extend(bytestream2_get_le16(&gb), 16);
                    previous_sample = sign_extend(bytestream2_get_le16(&gb), 16);
                }
                else
                {
                    current_sample  = c->status[channel].predictor;
                    previous_sample = c->status[channel].prev_sample;
                }

                for (count1 = 0; count1 < nb_samples / 28; count1++)
                {
                    int byte = bytestream2_get_byte(&gb);
                    if (byte == 0xEE)
                    {  /* only seen in R2 and R3 */
                        current_sample  = sign_extend(bytestream2_get_be16(&gb), 16);
                        previous_sample = sign_extend(bytestream2_get_be16(&gb), 16);

                        for (count2=0; count2<28; count2++)
                            *samplesC++ = sign_extend(bytestream2_get_be16(&gb), 16);
                    }
                    else
                    {
                        coeff1 = ea_adpcm_table[ byte >> 4];
                        coeff2 = ea_adpcm_table[(byte >> 4) + 4];
                        shift = 20 - (byte & 0x0F);

                        for (count2=0; count2<28; count2++)
                        {
                            if (count2 & 1)
                                next_sample = sign_extend(byte,    4) << shift;
                            else
                            {
                                byte = bytestream2_get_byte(&gb);
                                next_sample = sign_extend(byte >> 4, 4) << shift;
                            }

                            next_sample += (current_sample  * coeff1) +
                                           (previous_sample * coeff2);
                            next_sample = av_clip_int16_c(next_sample >> 8);

                            previous_sample = current_sample;
                            current_sample  = next_sample;
                            *samplesC++ = current_sample;
                        }
                    }
                }
                if (!count)
                {
                    count = count1;
                }
                else if (count != count1)
                {
                    //av_log(avctx, AV_LOG_WARNING, "per-channel sample count mismatch\n");
                    count = FFMAX(count, count1);
                }

                if (avctx->audio.codec_tag != AV_CODEC_ID_ADPCM_EA_R1)
                {
                    c->status[channel].predictor   = current_sample;
                    c->status[channel].prev_sample = previous_sample;
                }
            }

            output_buff_size = (avctx->audio.channels * avctx->audio.bits_per_sample * count * 28) / 8;
            bytestream2_seek(&gb, 0, SEEK_END);
            break;
        }
        case AV_CODEC_ID_ADPCM_EA_XAS:
            for (channel=0; channel<avctx->audio.channels; channel++)
            {
                int coeff[2][4], shift[4];
                int16_t *s = samples_p[channel];
                for (n = 0; n < 4; n++, s += 32)
                {
                    int val = sign_extend(bytestream2_get_le16u(&gb), 16);
                    for (i=0; i<2; i++)
                        coeff[i][n] = ea_adpcm_table[(val&0x0F)+4*i];
                    s[0] = val & ~0x0F;

                    val = sign_extend(bytestream2_get_le16u(&gb), 16);
                    shift[n] = 20 - (val & 0x0F);
                    s[1] = val & ~0x0F;
                }

                for (m=2; m<32; m+=2)
                {
                    s = &samples_p[channel][m];
                    for (n = 0; n < 4; n++, s += 32)
                    {
                        int level, pred;
                        int byte = bytestream2_get_byteu(&gb);

                        level = sign_extend(byte >> 4, 4) << shift[n];
                        pred  = s[-1] * coeff[0][n] + s[-2] * coeff[1][n];
                        s[0]  = av_clip_int16_c((level + pred + 0x80) >> 8);

                        level = sign_extend(byte, 4) << shift[n];
                        pred  = s[0] * coeff[0][n] + s[-1] * coeff[1][n];
                        s[1]  = av_clip_int16_c((level + pred + 0x80) >> 8);
                    }
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_AMV:
            c->status[0].predictor = sign_extend(bytestream2_get_le16u(&gb), 16);
            c->status[0].step_index = bytestream2_get_byteu(&gb);
            bytestream2_skipu(&gb, 5);
            if (c->status[0].step_index > 88u)
            {
                //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n",
                //       c->status[0].step_index);
                return -1;
            }

            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int v = bytestream2_get_byteu(&gb);

                *samples++ = adpcm_ima_expand_nibble(&c->status[0], v >> 4, 3);
                *samples++ = adpcm_ima_expand_nibble(&c->status[0], v & 0xf, 3);
            }
            break;
        case AV_CODEC_ID_ADPCM_IMA_SMJPEG:
            for (i = 0; i < avctx->audio.channels; i++)
            {
                c->status[i].predictor = sign_extend(bytestream2_get_be16u(&gb), 16);
                c->status[i].step_index = bytestream2_get_byteu(&gb);
                bytestream2_skipu(&gb, 1);
                if (c->status[i].step_index > 88u)
                {
                    //av_log(avctx, AV_LOG_ERROR, "ERROR: step_index = %i\n",
                    //       c->status[i].step_index);
                    return -1;
                }
            }

            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int v = bytestream2_get_byteu(&gb);

                *samples++ = adpcm_ima_qt_expand_nibble(&c->status[0 ], v >> 4, 3);
                *samples++ = adpcm_ima_qt_expand_nibble(&c->status[st], v & 0xf, 3);
            }
            break;
        case AV_CODEC_ID_ADPCM_CT:
            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int v = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4  );
                *samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F);
            }
            break;
        case AV_CODEC_ID_ADPCM_SBPRO_4:
        case AV_CODEC_ID_ADPCM_SBPRO_3:
        case AV_CODEC_ID_ADPCM_SBPRO_2:
            if (!c->status[0].step_index)
            {
                /* the first byte is a raw sample */
                *samples++ = 128 * (bytestream2_get_byteu(&gb) - 0x80);
                if (st)
                    *samples++ = 128 * (bytestream2_get_byteu(&gb) - 0x80);
                c->status[0].step_index = 1;
                nb_samples--;
            }
            if (avctx->audio.codec_tag == AV_CODEC_ID_ADPCM_SBPRO_4)
            {
                for (n = nb_samples >> (1 - st); n > 0; n--)
                {
                    int byte = bytestream2_get_byteu(&gb);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                           byte >> 4,   4, 0);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
                                                           byte & 0x0F, 4, 0);
                }
            }
            else if (avctx->audio.codec_tag == AV_CODEC_ID_ADPCM_SBPRO_3)
            {
                for (n = (nb_samples<<st) / 3; n > 0; n--)
                {
                    int byte = bytestream2_get_byteu(&gb);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                            byte >> 5        , 3, 0);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                           (byte >> 2) & 0x07, 3, 0);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                            byte & 0x03,       2, 0);
                }
            }
            else
            {
                for (n = nb_samples >> (2 - st); n > 0; n--)
                {
                    int byte = bytestream2_get_byteu(&gb);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                            byte >> 6        , 2, 2);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
                                                           (byte >> 4) & 0x03, 2, 2);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
                                                           (byte >> 2) & 0x03, 2, 2);
                    *samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
                                                            byte & 0x03,       2, 2);
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_SWF:
            adpcm_swf_decode(avctx, buf, buf_size, samples);
            bytestream2_seek(&gb, 0, SEEK_END);
            break;
        case AV_CODEC_ID_ADPCM_YAMAHA:
            for (n = nb_samples >> (1 - st); n > 0; n--)
            {
                int v = bytestream2_get_byteu(&gb);
                *samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F);
                *samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4  );
            }
            break;
        case AV_CODEC_ID_ADPCM_AICA:
            if (!c->has_status)
            {
                for (channel = 0; channel < avctx->audio.channels; channel++)
                    c->status[channel].step = 0;
                c->has_status = 1;
            }
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                samples = samples_p[channel];
                for (n = nb_samples >> 1; n > 0; n--)
                {
                    int v = bytestream2_get_byteu(&gb);
                    *samples++ = adpcm_yamaha_expand_nibble(&c->status[channel], v & 0x0F);
                    *samples++ = adpcm_yamaha_expand_nibble(&c->status[channel], v >> 4  );
                }
            }
            break;
        case AV_CODEC_ID_ADPCM_AFC:
        {
            int samples_per_block;
            int blocks;

            if (avctx->audio.extradata && avctx->audio.extradata_size == 1 && avctx->audio.extradata[0])
            {
                samples_per_block = avctx->audio.extradata[0] / 16;
                blocks = nb_samples / avctx->audio.extradata[0];
            }
            else
            {
                samples_per_block = nb_samples / 16;
                blocks = 1;
            }

            for (m = 0; m < blocks; m++)
            {
                for (channel = 0; channel < avctx->audio.channels; channel++)
                {
                    int prev1 = c->status[channel].sample1;
                    int prev2 = c->status[channel].sample2;

                    samples = samples_p[channel] + m * 16;
                    /* Read in every sample for this channel.  */
                    for (i = 0; i < samples_per_block; i++)
                    {
                        int byte = bytestream2_get_byteu(&gb);
                        int scale = 1 << (byte >> 4);
                        int index = byte & 0xf;
                        int factor1 = ff_adpcm_afc_coeffs[0][index];
                        int factor2 = ff_adpcm_afc_coeffs[1][index];

                        /* Decode 16 samples.  */
                        for (n = 0; n < 16; n++)
                        {
                            int32_t sampledat;

                            if (n & 1)
                            {
                                sampledat = sign_extend(byte, 4);
                            }
                            else
                            {
                                byte = bytestream2_get_byteu(&gb);
                                sampledat = sign_extend(byte >> 4, 4);
                            }

                            sampledat = ((prev1 * factor1 + prev2 * factor2) +
                                         ((sampledat * scale) << 11)) >> 11;
                            *samples = av_clip_int16_c(sampledat);
                            prev2 = prev1;
                            prev1 = *samples++;
                        }
                    }

                    c->status[channel].sample1 = prev1;
                    c->status[channel].sample2 = prev2;
                }
            }
            bytestream2_seek(&gb, 0, SEEK_END);
            break;
        }
        case AV_CODEC_ID_ADPCM_THP:
        case AV_CODEC_ID_ADPCM_THP_LE:
        {
            int table[14][16];
            int ch;

    #define THP_GET16(g) \
        sign_extend( \
            avctx->audio.codec_tag == AV_CODEC_ID_ADPCM_THP_LE ? \
            bytestream2_get_le16u(&(g)) : \
            bytestream2_get_be16u(&(g)), 16)

            if (avctx->audio.extradata)
            {
                GetByteContext tb;
                if (avctx->audio.extradata_size < 32 * avctx->audio.channels)
                {
                    //av_log(avctx, AV_LOG_ERROR, "Missing coeff table\n");
                    return -1;
                }

                bytestream2_init(&tb, avctx->audio.extradata, avctx->audio.extradata_size);
                for (i = 0; i < avctx->audio.channels; i++)
                    for (n = 0; n < 16; n++)
                        table[i][n] = THP_GET16(tb);
            }
            else
            {
                for (i = 0; i < avctx->audio.channels; i++)
                    for (n = 0; n < 16; n++)
                        table[i][n] = THP_GET16(gb);

                if (!c->has_status)
                {
                    /* Initialize the previous sample.  */
                    for (i = 0; i < avctx->audio.channels; i++)
                    {
                        c->status[i].sample1 = THP_GET16(gb);
                        c->status[i].sample2 = THP_GET16(gb);
                    }
                    c->has_status = 1;
                }
                else
                {
                    bytestream2_skip(&gb, avctx->audio.channels * 4);
                }
            }

            for (ch = 0; ch < avctx->audio.channels; ch++)
            {
                samples = samples_p[ch];

                /* Read in every sample for this channel.  */
                for (i = 0; i < (nb_samples + 13) / 14; i++)
                {
                    int byte = bytestream2_get_byteu(&gb);
                    int index = (byte >> 4) & 7;
                    unsigned int exp = byte & 0x0F;
                    int factor1 = table[ch][index * 2];
                    int factor2 = table[ch][index * 2 + 1];

                    /* Decode 14 samples.  */
                    for (n = 0; n < 14 && (i * 14 + n < nb_samples); n++)
                    {
                        int32_t sampledat;

                        if (n & 1)
                        {
                            sampledat = sign_extend(byte, 4);
                        }
                        else
                        {
                            byte = bytestream2_get_byteu(&gb);
                            sampledat = sign_extend(byte >> 4, 4);
                        }

                        sampledat = ((c->status[ch].sample1 * factor1
                                    + c->status[ch].sample2 * factor2) >> 11) + (sampledat << exp);
                        *samples = av_clip_int16_c(sampledat);
                        c->status[ch].sample2 = c->status[ch].sample1;
                        c->status[ch].sample1 = *samples++;
                    }
                }
            }
            break;
        }
        case AV_CODEC_ID_ADPCM_DTK:
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                samples = samples_p[channel];

                /* Read in every sample for this channel.  */
                for (i = 0; i < nb_samples / 28; i++)
                {
                    int byte, header;
                    if (channel)
                        bytestream2_skipu(&gb, 1);
                    header = bytestream2_get_byteu(&gb);
                    bytestream2_skipu(&gb, 3 - channel);

                    /* Decode 28 samples.  */
                    for (n = 0; n < 28; n++)
                    {
                        int32_t sampledat, prev;

                        switch (header >> 4)
                        {
                        case 1:
                            prev = (c->status[channel].sample1 * 0x3c);
                            break;
                        case 2:
                            prev = (c->status[channel].sample1 * 0x73) - (c->status[channel].sample2 * 0x34);
                            break;
                        case 3:
                            prev = (c->status[channel].sample1 * 0x62) - (c->status[channel].sample2 * 0x37);
                            break;
                        default:
                            prev = 0;
                        }

                        prev = av_clip_intp2_c((prev + 0x20) >> 6, 21);

                        byte = bytestream2_get_byteu(&gb);
                        if (!channel)
                            sampledat = sign_extend(byte, 4);
                        else
                            sampledat = sign_extend(byte >> 4, 4);

                        sampledat = (((sampledat << 12) >> (header & 0xf)) << 6) + prev;
                        *samples++ = av_clip_int16_c(sampledat >> 6);
                        c->status[channel].sample2 = c->status[channel].sample1;
                        c->status[channel].sample1 = sampledat;
                    }
                }
                if (!channel)
                    bytestream2_seek(&gb, 0, SEEK_SET);
            }
            break;
        case AV_CODEC_ID_ADPCM_PSX:
            for (channel = 0; channel < avctx->audio.channels; channel++)
            {
                samples = samples_p[channel];

                /* Read in every sample for this channel.  */
                for (i = 0; i < nb_samples / 28; i++)
                {
                    int filter, shift, flag, byte;

                    filter = bytestream2_get_byteu(&gb);
                    shift  = filter & 0xf;
                    filter = filter >> 4;
                    if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table))
                        return -1;
                    flag   = bytestream2_get_byteu(&gb);

                    /* Decode 28 samples.  */
                    byte = 0;
                    for (n = 0; n < 28; n++)
                    {
                        int sample = 0, scale;
                        if (flag < 0x07)
                        {
                            if (n & 1)
                            {
                                scale = sign_extend(byte >> 4, 4);
                            }
                            else
                            {
                                byte  = bytestream2_get_byteu(&gb);
                                scale = sign_extend(byte, 4);
                            }

                            scale  = scale << 12;
                            sample = (int)((scale >> shift) + (c->status[channel].sample1 * xa_adpcm_table[filter][0] + c->status[channel].sample2 * xa_adpcm_table[filter][1]) / 64);
                        }
                        *samples++ = av_clip_int16_c(sample);
                        c->status[channel].sample2 = c->status[channel].sample1;
                        c->status[channel].sample1 = sample;
                    }
                }
            }
            break;

        default:
            return -1;
    }

    if (avpkt->size && bytestream2_tell(&gb) == 0)
    {
        //av_log(avctx, AV_LOG_ERROR, "Nothing consumed\n");
        return -1;
    }

    if (avpkt->size < bytestream2_tell(&gb))
    {
        //av_log(avctx, AV_LOG_ERROR, "Overread of %d < %d\n", avpkt->size, bytestream2_tell(&gb));
        return avpkt->size;
    }

    avctx->audio.buff_offset += avctx->audio.buff_size;
    avctx->audio.buff_size = output_buff_size;

    return bytestream2_tell(&gb);
}


static void adpcm_free_data(void *data)
{
    ADPCMDecodeContext *c = (ADPCMDecodeContext*)data;
    if(c)
    {
        free(c);
    }
}

void adpcm_decode_init(struct tiny_codec_s *avctx)
{
    if(!avctx->audio.priv_data)
    {
        ADPCMDecodeContext *c = (ADPCMDecodeContext*)malloc(sizeof(ADPCMDecodeContext));
        unsigned int min_channels = 1;
        unsigned int max_channels = 2;

        avctx->audio.priv_data = c;
        avctx->audio.free_data = adpcm_free_data;
        avctx->audio.decode = adpcm_decode_frame;

        switch(avctx->audio.codec_tag)
        {
            case AV_CODEC_ID_ADPCM_DTK:
            case AV_CODEC_ID_ADPCM_EA:
                min_channels = 2;
                break;
            case AV_CODEC_ID_ADPCM_AFC:
            case AV_CODEC_ID_ADPCM_EA_R1:
            case AV_CODEC_ID_ADPCM_EA_R2:
            case AV_CODEC_ID_ADPCM_EA_R3:
            case AV_CODEC_ID_ADPCM_EA_XAS:
                max_channels = 6;
                break;
            case AV_CODEC_ID_ADPCM_MTAF:
                min_channels = 2;
                max_channels = 8;
                break;
            case AV_CODEC_ID_ADPCM_PSX:
                max_channels = 8;
                break;
            case AV_CODEC_ID_ADPCM_IMA_DAT4:
            case AV_CODEC_ID_ADPCM_THP:
            case AV_CODEC_ID_ADPCM_THP_LE:
                max_channels = 14;
                break;
        }

        if (avctx->audio.channels < min_channels || avctx->audio.channels > max_channels)
        {
            free(avctx->audio.priv_data);
            avctx->audio.priv_data = NULL;
            avctx->audio.free_data = NULL;
            return;
        }

        switch(avctx->audio.codec_tag)
        {
            case AV_CODEC_ID_ADPCM_CT:
                c->status[0].step = c->status[1].step = 511;
                break;

            case AV_CODEC_ID_ADPCM_IMA_WAV:
                if (avctx->audio.bits_per_coded_sample < 2 || avctx->audio.bits_per_coded_sample > 5)
                {
                    free(avctx->audio.priv_data);
                    avctx->audio.priv_data = NULL;
                    avctx->audio.free_data = NULL;
                    return;
                }
                break;

            case AV_CODEC_ID_ADPCM_IMA_APC:
                if (avctx->audio.extradata && avctx->audio.extradata_size >= 8)
                {
                    c->status[0].predictor = AV_RL32(avctx->audio.extradata);
                    c->status[1].predictor = AV_RL32(avctx->audio.extradata + 4);
                }
                break;

            case AV_CODEC_ID_ADPCM_IMA_WS:
                if (avctx->audio.extradata && avctx->audio.extradata_size >= 2)
                    c->vqa_version = AV_RL16(avctx->audio.extradata);
                break;

            default:
                break;
        }

        switch(avctx->audio.codec_tag)
        {
            case AV_CODEC_ID_ADPCM_AICA:
            case AV_CODEC_ID_ADPCM_IMA_DAT4:
            case AV_CODEC_ID_ADPCM_IMA_QT:
            case AV_CODEC_ID_ADPCM_IMA_WAV:
            case AV_CODEC_ID_ADPCM_4XM:
            case AV_CODEC_ID_ADPCM_XA:
            case AV_CODEC_ID_ADPCM_EA_R1:
            case AV_CODEC_ID_ADPCM_EA_R2:
            case AV_CODEC_ID_ADPCM_EA_R3:
            case AV_CODEC_ID_ADPCM_EA_XAS:
            case AV_CODEC_ID_ADPCM_THP:
            case AV_CODEC_ID_ADPCM_THP_LE:
            case AV_CODEC_ID_ADPCM_AFC:
            case AV_CODEC_ID_ADPCM_DTK:
            case AV_CODEC_ID_ADPCM_PSX:
            case AV_CODEC_ID_ADPCM_MTAF:
                //avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
                avctx->audio.bits_per_sample = 16;
                break;
            case AV_CODEC_ID_ADPCM_IMA_WS:
                /*avctx->sample_fmt = c->vqa_version == 3 ? AV_SAMPLE_FMT_S16P :
                                                          AV_SAMPLE_FMT_S16;*/
                avctx->audio.bits_per_sample = 16;
                break;
            default:
                //avctx->sample_fmt = AV_SAMPLE_FMT_S16;
                avctx->audio.bits_per_sample = 16;
        }
    }
}
